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590408dca3
When we receive a SIP reinvite, it is possible that we may not be able to process the reinvite immediately since we have also sent a reinvite out ourselves. The problem is that whoever sent us the reinvite may have also sent a reinvite out to another party, and that reinvite may have succeeded. As a result, even though we are not going to accept the reinvite we just received, it is important for us to not have problems if we suddenly start receiving RTP from a new source. The fix for this is to grab the media source information from the SDP of the reinvite that we receive. This information is passed to the RTP layer so that it will know about the alternate source for media. Review: https://reviewboard.asterisk.org/r/252 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@197588 65c4cc65-6c06-0410-ace0-fbb531ad65f3
283 lines
9.5 KiB
C
283 lines
9.5 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 1999 - 2006, Digium, Inc.
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*
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* Mark Spencer <markster@digium.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*!
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* \file rtp.h
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* \brief Supports RTP and RTCP with Symmetric RTP support for NAT traversal.
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*
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* RTP is defined in RFC 3550.
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*/
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#ifndef _ASTERISK_RTP_H
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#define _ASTERISK_RTP_H
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#include <netinet/in.h>
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#include "asterisk/frame.h"
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#include "asterisk/io.h"
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#include "asterisk/sched.h"
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#include "asterisk/channel.h"
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#include "asterisk/linkedlists.h"
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#if defined(__cplusplus) || defined(c_plusplus)
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extern "C" {
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#endif
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/* Codes for RTP-specific data - not defined by our AST_FORMAT codes */
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/*! DTMF (RFC2833) */
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#define AST_RTP_DTMF (1 << 0)
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/*! 'Comfort Noise' (RFC3389) */
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#define AST_RTP_CN (1 << 1)
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/*! DTMF (Cisco Proprietary) */
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#define AST_RTP_CISCO_DTMF (1 << 2)
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/*! Maximum RTP-specific code */
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#define AST_RTP_MAX AST_RTP_CISCO_DTMF
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#define MAX_RTP_PT 256
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enum ast_rtp_options {
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AST_RTP_OPT_G726_NONSTANDARD = (1 << 0),
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};
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enum ast_rtp_get_result {
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/*! Failed to find the RTP structure */
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AST_RTP_GET_FAILED = 0,
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/*! RTP structure exists but true native bridge can not occur so try partial */
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AST_RTP_TRY_PARTIAL,
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/*! RTP structure exists and native bridge can occur */
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AST_RTP_TRY_NATIVE,
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};
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struct ast_rtp;
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struct ast_rtp_protocol {
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/*! Get RTP struct, or NULL if unwilling to transfer */
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enum ast_rtp_get_result (* const get_rtp_info)(struct ast_channel *chan, struct ast_rtp **rtp);
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/*! Get RTP struct, or NULL if unwilling to transfer */
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enum ast_rtp_get_result (* const get_vrtp_info)(struct ast_channel *chan, struct ast_rtp **rtp);
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/*! Set RTP peer */
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int (* const set_rtp_peer)(struct ast_channel *chan, struct ast_rtp *peer, struct ast_rtp *vpeer, int codecs, int nat_active);
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int (* const get_codec)(struct ast_channel *chan);
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const char * const type;
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AST_LIST_ENTRY(ast_rtp_protocol) list;
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};
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struct ast_rtp_quality {
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unsigned int local_ssrc; /* Our SSRC */
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unsigned int local_lostpackets; /* Our lost packets */
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double local_jitter; /* Our calculated jitter */
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unsigned int local_count; /* Number of received packets */
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unsigned int remote_ssrc; /* Their SSRC */
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unsigned int remote_lostpackets; /* Their lost packets */
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double remote_jitter; /* Their reported jitter */
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unsigned int remote_count; /* Number of transmitted packets */
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double rtt; /* Round trip time */
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};
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#define FLAG_3389_WARNING (1 << 0)
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typedef int (*ast_rtp_callback)(struct ast_rtp *rtp, struct ast_frame *f, void *data);
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/*!
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* \brief Get the amount of space required to hold an RTP session
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* \return number of bytes required
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*/
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size_t ast_rtp_alloc_size(void);
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/*!
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* \brief Initializate a RTP session.
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*
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* \param sched
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* \param io
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* \param rtcpenable
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* \param callbackmode
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* \returns A representation (structure) of an RTP session.
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*/
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struct ast_rtp *ast_rtp_new(struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode);
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/*!
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* \brief Initializate a RTP session using an in_addr structure.
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*
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* This fuction gets called by ast_rtp_new().
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*
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* \param sched
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* \param io
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* \param rtcpenable
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* \param callbackmode
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* \param in
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* \returns A representation (structure) of an RTP session.
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*/
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struct ast_rtp *ast_rtp_new_with_bindaddr(struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode, struct in_addr in);
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void ast_rtp_set_peer(struct ast_rtp *rtp, struct sockaddr_in *them);
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/*!
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* \since 1.4.26
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* \brief set potential alternate source for RTP media
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*
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* This function may be used to give the RTP stack a hint that there is a potential
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* second source of media. One case where this is used is when the SIP stack receives
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* a REINVITE to which it will be replying with a 491. In such a scenario, the IP and
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* port information in the SDP of that REINVITE lets us know that we may receive media
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* from that source/those sources even though the SIP transaction was unable to be completed
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* successfully
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*
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* \param rtp The RTP structure we wish to set up an alternate host/port on
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* \param alt The address information for the alternate media source
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* \retval void
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*/
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void ast_rtp_set_alt_peer(struct ast_rtp *rtp, struct sockaddr_in *alt);
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/* Copies from rtp to them and returns 1 if there was a change or 0 if it was already the same */
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int ast_rtp_get_peer(struct ast_rtp *rtp, struct sockaddr_in *them);
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void ast_rtp_get_us(struct ast_rtp *rtp, struct sockaddr_in *us);
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struct ast_rtp *ast_rtp_get_bridged(struct ast_rtp *rtp);
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void ast_rtp_destroy(struct ast_rtp *rtp);
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void ast_rtp_reset(struct ast_rtp *rtp);
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void ast_rtp_stun_request(struct ast_rtp *rtp, struct sockaddr_in *suggestion, const char *username);
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void ast_rtp_set_callback(struct ast_rtp *rtp, ast_rtp_callback callback);
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void ast_rtp_set_data(struct ast_rtp *rtp, void *data);
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int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *f);
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struct ast_frame *ast_rtp_read(struct ast_rtp *rtp);
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struct ast_frame *ast_rtcp_read(struct ast_rtp *rtp);
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int ast_rtp_fd(struct ast_rtp *rtp);
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int ast_rtcp_fd(struct ast_rtp *rtp);
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int ast_rtp_senddigit_begin(struct ast_rtp *rtp, char digit);
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int ast_rtp_senddigit_end(struct ast_rtp *rtp, char digit);
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int ast_rtp_sendcng(struct ast_rtp *rtp, int level);
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int ast_rtp_settos(struct ast_rtp *rtp, int tos);
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void ast_rtp_new_source(struct ast_rtp *rtp);
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/*! \brief Setting RTP payload types from lines in a SDP description: */
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void ast_rtp_pt_clear(struct ast_rtp* rtp);
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/*! \brief Set payload types to defaults */
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void ast_rtp_pt_default(struct ast_rtp* rtp);
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/*! \brief Copy payload types between RTP structures */
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void ast_rtp_pt_copy(struct ast_rtp *dest, struct ast_rtp *src);
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/*! \brief Activate payload type */
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void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt);
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/*! \brief clear payload type */
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void ast_rtp_unset_m_type(struct ast_rtp* rtp, int pt);
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/*! \brief Initiate payload type to a known MIME media type for a codec */
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int ast_rtp_set_rtpmap_type(struct ast_rtp* rtp, int pt,
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char *mimeType, char *mimeSubtype,
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enum ast_rtp_options options);
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/*! \brief Mapping between RTP payload format codes and Asterisk codes: */
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struct rtpPayloadType ast_rtp_lookup_pt(struct ast_rtp* rtp, int pt);
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int ast_rtp_lookup_code(struct ast_rtp* rtp, int isAstFormat, int code);
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void ast_rtp_get_current_formats(struct ast_rtp* rtp,
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int* astFormats, int* nonAstFormats);
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/*! \brief Mapping an Asterisk code into a MIME subtype (string): */
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const char *ast_rtp_lookup_mime_subtype(int isAstFormat, int code,
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enum ast_rtp_options options);
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/*! \brief Build a string of MIME subtype names from a capability list */
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char *ast_rtp_lookup_mime_multiple(char *buf, size_t size, const int capability,
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const int isAstFormat, enum ast_rtp_options options);
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void ast_rtp_setnat(struct ast_rtp *rtp, int nat);
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int ast_rtp_getnat(struct ast_rtp *rtp);
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/*! \brief Indicate whether this RTP session is carrying DTMF or not */
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void ast_rtp_setdtmf(struct ast_rtp *rtp, int dtmf);
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/*! \brief Compensate for devices that send RFC2833 packets all at once */
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void ast_rtp_setdtmfcompensate(struct ast_rtp *rtp, int compensate);
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/*! \brief Enable STUN capability */
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void ast_rtp_setstun(struct ast_rtp *rtp, int stun_enable);
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int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms);
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int ast_rtp_proto_register(struct ast_rtp_protocol *proto);
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void ast_rtp_proto_unregister(struct ast_rtp_protocol *proto);
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int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, int media);
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/*! \brief If possible, create an early bridge directly between the devices without
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having to send a re-invite later */
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int ast_rtp_early_bridge(struct ast_channel *dest, struct ast_channel *src);
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void ast_rtp_stop(struct ast_rtp *rtp);
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/*! \brief Return RTCP quality string */
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char *ast_rtp_get_quality(struct ast_rtp *rtp, struct ast_rtp_quality *qual);
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/*! \brief Send an H.261 fast update request. Some devices need this rather than the XML message in SIP */
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int ast_rtcp_send_h261fur(void *data);
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void ast_rtp_new_init(struct ast_rtp *rtp);
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void ast_rtp_init(void);
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int ast_rtp_reload(void);
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int ast_rtp_codec_setpref(struct ast_rtp *rtp, struct ast_codec_pref *prefs);
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struct ast_codec_pref *ast_rtp_codec_getpref(struct ast_rtp *rtp);
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int ast_rtp_codec_getformat(int pt);
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/*! \brief Set rtp timeout */
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void ast_rtp_set_rtptimeout(struct ast_rtp *rtp, int timeout);
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/*! \brief Set rtp hold timeout */
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void ast_rtp_set_rtpholdtimeout(struct ast_rtp *rtp, int timeout);
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/*! \brief set RTP keepalive interval */
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void ast_rtp_set_rtpkeepalive(struct ast_rtp *rtp, int period);
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/*! \brief Get RTP keepalive interval */
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int ast_rtp_get_rtpkeepalive(struct ast_rtp *rtp);
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/*! \brief Get rtp hold timeout */
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int ast_rtp_get_rtpholdtimeout(struct ast_rtp *rtp);
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/*! \brief Get rtp timeout */
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int ast_rtp_get_rtptimeout(struct ast_rtp *rtp);
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/* \brief Put RTP timeout timers on hold during another transaction, like T.38 */
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void ast_rtp_set_rtptimers_onhold(struct ast_rtp *rtp);
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#if defined(__cplusplus) || defined(c_plusplus)
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}
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#endif
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#endif /* _ASTERISK_RTP_H */
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